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Call-Back Rules

ciscojoe837
Level 1
Level 1

I'm trying to configure a UC540W (latest CCA and SWP) with call-back rules but all I get is "Unknow Number" when I try to dial a missed number -- it never puts the 9 in there.

I'm looking for the CLI changes that are suppose to take place for this feature to work so I can figure out what the CCA didn't do.

I'm using both PSTN and SIP trunks.

Thanks.

6 Replies 6

David Trad
VIP Alumni
VIP Alumni

Hey There Joe,

Can you take a screen shot of what you have in CCA for the call back rules?

Also how many numbers does the ITSP require for the SIP trunk, and how many numbers are required for the PSTN carrier... All this needs to be taken into consideration.

If you want to know what changes CCA makes, you need to read the dialogue box that appears after you have applied the changes, alternatively you can press F2 and this will bring up a debugging screen and you can see what is taking place in there which is also quite handy.

Cheers,

David.

Cheers, David Trad. **When you rate a persons post, you are indicating a thank you or that it helped, but at the same time you are also helping to maintain the community spirit - You don't have to rate posts and you wont be looked down upon :) *

I'm familiar with what got changed so that part is not a problem.  I would like to know what the CLI changes are from someone that has it working.  If someone has CLI logging enabled in CCA they could cut and paste what got changed here.  I can figure it out from there.  I want to see from a working config what the translation rule is and where it gets applied.

Thanks.

lgaughan
Level 1
Level 1

If you are trying to get this to work over FXOs, CCA does not handle them currently.  This is a known issue that will be fixed in the very next release.  Is this your case?

For SIP Trunk, this should be working as long as you haved also defined your Incoming DID Dial Plans with CCA.   Call-back rules are written to translation-rule 3265.  This rule is then added to the translation-profiles associated with the inbound dial-peers (3xxx range) for the calling number.

Laura

Not working with FXO is an important point that I don't remember seeing anywhere.  You would think there would be a huge note in red DOES NOT WORK WITH FXO, SIP ONLY!  I don't get how this point doesn't get mentioned.

Having said that, I've tried going out the SIP trunks only and it still doesn't work.  Yes, it created the 3265 translation rules but they are not applied anywhere (actually it's under a VM_Profile which I don't get at all).  So that's why I wanted to see a working example of a translation profile and then dial-peer. 

I still don't get how this even made it out of QA in the first place.

Thanks.

Yeah, I agree with you about FXO.  It was *supposed* to be there, which is why it was not documented as a major limitation.  The probelms are being addressed in the next release. 

On the PSTN, it has been confirmed to be working by many using ISDN (PRIs and BRIs).

For SIP,  were your Incoming Dial Plans configured with 3.1.1 or a previous version of CCA?  In the first case, all translation-profiles get this rule applied by default, whether or not any call-back rules are actually defined.  In the second case, there may be an issue with CCA detecting and applying it to older dial plans. 

If you can attach the CCA troubleshooting log (or at least a running config), that would be helpful to start getting this narrowed down.  If you are uncomfortable doing that here, perhaps open a case and post back the number so I can take a look.

Laura

It was just the previous version of 3.1.1.

I didn't know it was suppose to work with the incoming dial plans so thanks for that.  I thought the access code (say, 9) would be added when dialing out.

For me, all the CCA did was add the translation rule 3265 and applied the rule to a VM translation profile -- not any of the incoming dial peers.

That said,  I just manually added Translate Calling 3265 to the translation profile of the incoming sip dial-peers that hit the auto attendant.

I then added a Translation Profile with Translation Rule 3265 and put it on the dial peer on the FXO port (in this case I only have one).

Now, both SIP and PSTN incoming caller-id numbers get the 9 added so any missed calls can go out immediately withouth editing the number and putting in the access code.

Why was putting it on FXO dial peers such a big deal?  I'm still in disbelief this was released through CCA.  It was so easy for it not to work.  I would have thought more users would have come across the problem.